Merge branch 'planar_audio'

Conflicts:
	audio/out/ao_lavc.c

65 files changed, 1549 insertions, 1489 deletions

diff --git a/Makefile b/Makefile
index 771b2a6245..495fe79e61 100644
--- a/Makefile
+++ b/Makefile
@@ -139,6 +139,7 @@ endif
 SOURCES-$(DLOPEN)               += video/filter/vf_dlopen.c
 
 SOURCES = audio/audio.c \
+          audio/audio_buffer.c \
           audio/chmap.c \
           audio/chmap_sel.c \
           audio/fmt-conversion.c \
diff --git a/audio/audio.c b/audio/audio.c
index 9d41928436..2a67e22e5e 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -17,42 +17,60 @@
 
 #include <assert.h>
 
+#include <libavutil/mem.h>
+
+#include "mpvcore/mp_common.h"
 #include "mpvcore/mp_talloc.h"
 #include "audio.h"
 
+static void update_redundant_info(struct mp_audio *mpa)
+{
+    assert(mp_chmap_is_empty(&mpa->channels) ||
+           mp_chmap_is_valid(&mpa->channels));
+    mpa->nch = mpa->channels.num;
+    mpa->bps = af_fmt2bits(mpa->format) / 8;
+    if (af_fmt_is_planar(mpa->format)) {
+        mpa->spf = 1;
+        mpa->num_planes = mpa->nch;
+        mpa->sstride = mpa->bps;
+    } else {
+        mpa->spf = mpa->nch;
+        mpa->num_planes = 1;
+        mpa->sstride = mpa->bps * mpa->nch;
+    }
+}
+
 void mp_audio_set_format(struct mp_audio *mpa, int format)
 {
     mpa->format = format;
-    mpa->bps = af_fmt2bits(format) / 8;
+    update_redundant_info(mpa);
 }
 
 void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels)
 {
-    struct mp_chmap map;
-    mp_chmap_from_channels(&map, num_channels);
-    mp_audio_set_channels(mpa, &map);
+    mp_chmap_from_channels(&mpa->channels, num_channels);
+    update_redundant_info(mpa);
 }
 
 // Use old MPlayer/ALSA channel layout.
 void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels)
 {
-    struct mp_chmap map;
-    mp_chmap_from_channels_alsa(&map, num_channels);
-    mp_audio_set_channels(mpa, &map);
+    mp_chmap_from_channels_alsa(&mpa->channels, num_channels);
+    update_redundant_info(mpa);
 }
 
 void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap)
 {
-    assert(mp_chmap_is_empty(chmap) || mp_chmap_is_valid(chmap));
     mpa->channels = *chmap;
-    mpa->nch = mpa->channels.num;
+    update_redundant_info(mpa);
 }
 
 void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src)
 {
-    mp_audio_set_format(dst, src->format);
-    mp_audio_set_channels(dst, &src->channels);
+    dst->format = src->format;
+    dst->channels = src->channels;
     dst->rate = src->rate;
+    update_redundant_info(dst);
 }
 
 bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b)
@@ -74,3 +92,135 @@ char *mp_audio_config_to_str(struct mp_audio *mpa)
 {
     return mp_audio_fmt_to_str(mpa->rate, &mpa->channels, mpa->format);
 }
+
+void mp_audio_force_interleaved_format(struct mp_audio *mpa)
+{
+    if (af_fmt_is_planar(mpa->format))
+        mp_audio_set_format(mpa, af_fmt_from_planar(mpa->format));
+}
+
+// Return used size of a plane. (The size is the same for all planes.)
+int mp_audio_psize(struct mp_audio *mpa)
+{
+    return mpa->samples * mpa->sstride;
+}
+
+void mp_audio_set_null_data(struct mp_audio *mpa)
+{
+    for (int n = 0; n < MP_NUM_CHANNELS; n++)
+        mpa->planes[n] = NULL;
+    mpa->samples = 0;
+}
+
+static void mp_audio_destructor(void *ptr)
+{
+    struct mp_audio *mpa = ptr;
+    for (int n = mpa->num_planes; n < MP_NUM_CHANNELS; n++) {
+        // Note: don't free if not allocated by mp_audio_realloc
+        if (mpa->allocated[n])
+            av_free(mpa->planes[n]);
+    }
+}
+
+/* Reallocate the data stored in mpa->planes[n] so that enough samples are
+ * available on every plane. The previous data is kept (for the smallest
+ * common number of samples before/after resize).
+ *
+ * mpa->samples is not set or used.
+ *
+ * This function is flexible enough to handle format and channel layout
+ * changes. In these cases, all planes are reallocated as needed. Unused
+ * planes are freed.
+ *
+ * mp_audio_realloc(mpa, 0) will still yield non-NULL for mpa->data[n].
+ *
+ * Allocated data is implicitly freed on talloc_free(mpa).
+ */
+void mp_audio_realloc(struct mp_audio *mpa, int samples)
+{
+    assert(samples >= 0);
+    int size = MPMAX(samples * mpa->sstride, 1);
+    for (int n = 0; n < mpa->num_planes; n++) {
+        if (size != mpa->allocated[n]) {
+            // Note: av_realloc() can't be used (see libavutil doxygen)
+            void *new = av_malloc(size);
+            if (!new)
+                abort();
+            if (mpa->allocated[n])
+                memcpy(new, mpa->planes[n], MPMIN(mpa->allocated[n], size));
+            av_free(mpa->planes[n]);
+            mpa->planes[n] = new;
+            mpa->allocated[n] = size;
+        }
+    }
+    for (int n = mpa->num_planes; n < MP_NUM_CHANNELS; n++) {
+        av_free(mpa->planes[n]);
+        mpa->planes[n] = NULL;
+        mpa->allocated[n] = 0;
+    }
+    talloc_set_destructor(mpa, mp_audio_destructor);
+}
+
+// Like mp_audio_realloc(), but only reallocate if the audio grows in size.
+void mp_audio_realloc_min(struct mp_audio *mpa, int samples)
+{
+    if (samples > mp_audio_get_allocated_size(mpa))
+        mp_audio_realloc(mpa, samples);
+}
+
+/* Get the size allocated for the data, in number of samples. If the allocated
+ * size isn't on sample boundaries (e.g. after format changes), the returned
+ * sample number is a rounded down value.
+ *
+ * Note that this only works in situations where mp_audio_realloc() also works!
+ */
+int mp_audio_get_allocated_size(struct mp_audio *mpa)
+{
+    int size = 0;
+    for (int n = 0; n < mpa->num_planes; n++) {
+        int s = mpa->allocated[n] / mpa->sstride;
+        size = n == 0 ? s : MPMIN(size, s);
+    }
+    return size;
+}
+
+// Clear the samples [start, start + length) with silence.
+void mp_audio_fill_silence(struct mp_audio *mpa, int start, int length)
+{
+    assert(start >= 0 && length >= 0 && start + length <= mpa->samples);
+    int offset = start * mpa->sstride;
+    int size = length * mpa->sstride;
+    for (int n = 0; n < mpa->num_planes; n++) {
+        if (n > 0 && mpa->planes[n] == mpa->planes[0])
+            continue; // silly optimization for special cases
+        af_fill_silence((char *)mpa->planes[n] + offset, size, mpa->format);
+    }
+}
+
+// All integer parameters are in samples.
+// dst and src can overlap.
+void mp_audio_copy(struct mp_audio *dst, int dst_offset,
+                   struct mp_audio *src, int src_offset, int length)
+{
+    assert(mp_audio_config_equals(dst, src));
+    assert(length >= 0);
+    assert(dst_offset >= 0 && dst_offset + length <= dst->samples);
+    assert(src_offset >= 0 && src_offset + length <= src->samples);
+
+    for (int n = 0; n < dst->num_planes; n++) {
+        memmove((char *)dst->planes[n] + dst_offset * dst->sstride,
+                (char *)src->planes[n] + src_offset * src->sstride,
+                length * dst->sstride);
+    }
+}
+
+// Set data to the audio after the given number of samples (i.e. slice it).
+void mp_audio_skip_samples(struct mp_audio *data, int samples)
+{
+    assert(samples >= 0 && samples <= data->samples);
+
+    for (int n = 0; n < data->num_planes; n++)
+        data->planes[n] = (uint8_t *)data->planes[n] + samples * data->sstride;
+
+    data->samples -= samples;
+}
diff --git a/audio/audio.h b/audio/audio.h
index de35e697c8..54ac2d5aac 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -23,14 +23,22 @@
 
 // Audio data chunk
 struct mp_audio {
-    void *audio; // data buffer
-    int len;    // buffer length (in bytes)
-    int rate;   // sample rate
+    int samples;        // number of samples in data (per channel)
+    void *planes[MP_NUM_CHANNELS]; // data buffer (one per plane)
+    int rate;           // sample rate
     struct mp_chmap channels; // channel layout, use mp_audio_set_*() to set
     int format; // format (AF_FORMAT_...), use mp_audio_set_format() to set
     // Redundant fields, for convenience
-    int nch;    // number of channels (redundant with chmap)
-    int bps;    // bytes per sample (redundant with format)
+    int sstride;        // distance between 2 samples in bytes on a plane
+                        //  interleaved: bps * nch
+                        //  planar:      bps
+    int nch;            // number of channels (redundant with chmap)
+    int spf;            // sub-samples per sample on each plane
+    int num_planes;     // number of planes
+    int bps;            // size of sub-samples (af_fmt2bits(format) / 8)
+
+    // private
+    int allocated[MP_NUM_CHANNELS]; // use mp_audio_get_allocated_size()
 };
 
 void mp_audio_set_format(struct mp_audio *mpa, int format);
@@ -43,4 +51,20 @@ bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b);
 char *mp_audio_fmt_to_str(int srate, const struct mp_chmap *chmap, int format);
 char *mp_audio_config_to_str(struct mp_audio *mpa);
 
+void mp_audio_force_interleaved_format(struct mp_audio *mpa);
+
+int mp_audio_psize(struct mp_audio *mpa);
+
+void mp_audio_set_null_data(struct mp_audio *mpa);
+
+void mp_audio_realloc(struct mp_audio *mpa, int samples);
+void mp_audio_realloc_min(struct mp_audio *mpa, int samples);
+int mp_audio_get_allocated_size(struct mp_audio *mpa);
+
+void mp_audio_fill_silence(struct mp_audio *mpa, int start, int length);
+
+void mp_audio_copy(struct mp_audio *dst, int dst_offset,
+                   struct mp_audio *src, int src_offset, int length);
+void mp_audio_skip_samples(struct mp_audio *data, int samples);
+
 #endif
diff --git a/audio/audio_buffer.c b/audio/audio_buffer.c
new file mode 100644
index 0000000000..53563f90c6
--- /dev/null
+++ b/audio/audio_buffer.c
@@ -0,0 +1,152 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stddef.h>
+#include <limits.h>
+#include <assert.h>
+
+#include "mpvcore/mp_common.h"
+
+#include "audio_buffer.h"
+#include "audio.h"
+#include "format.h"
+
+struct mp_audio_buffer {
+    struct mp_audio *buffer;
+};
+
+struct mp_audio_buffer *mp_audio_buffer_create(void *talloc_ctx)
+{
+    struct mp_audio_buffer *ab = talloc(talloc_ctx, struct mp_audio_buffer);
+    *ab = (struct mp_audio_buffer) {
+        .buffer = talloc_zero(ab, struct mp_audio),
+    };
+    return ab;
+}
+
+// Reinitialize the buffer, set a new format, drop old data.
+// The audio data in fmt is not used, only the format.
+void mp_audio_buffer_reinit(struct mp_audio_buffer *ab, struct mp_audio *fmt)
+{
+    mp_audio_copy_config(ab->buffer, fmt);
+    mp_audio_realloc(ab->buffer, 1);
+    ab->buffer->samples = 0;
+}
+
+void mp_audio_buffer_reinit_fmt(struct mp_audio_buffer *ab, int format,
+                                const struct mp_chmap *channels, int srate)
+{
+    struct mp_audio mpa = {0};
+    mp_audio_set_format(&mpa, format);
+    mp_audio_set_channels(&mpa, channels);
+    mpa.rate = srate;
+    mp_audio_buffer_reinit(ab, &mpa);
+}
+
+void mp_audio_buffer_get_format(struct mp_audio_buffer *ab,
+                                struct mp_audio *out_fmt)
+{
+    *out_fmt = (struct mp_audio){0};
+    mp_audio_copy_config(out_fmt, ab->buffer);
+}
+
+// Make the total size of the internal buffer at least this number of samples.
+void mp_audio_buffer_preallocate_min(struct mp_audio_buffer *ab, int samples)
+{
+    mp_audio_realloc_min(ab->buffer, samples);
+}
+
+// Get number of samples that can be written without forcing a resize of the
+// internal buffer.
+int mp_audio_buffer_get_write_available(struct mp_audio_buffer *ab)
+{
+    return mp_audio_get_allocated_size(ab->buffer) - ab->buffer->samples;
+}
+
+// Get a pointer to the end of the buffer (where writing would append). If the
+// internal buffer is too small for the given number of samples, it's resized.
+// After writing to the buffer, mp_audio_buffer_finish_write() has to be used
+// to make the written data part of the readable buffer.
+void mp_audio_buffer_get_write_buffer(struct mp_audio_buffer *ab, int samples,
+                                      struct mp_audio *out_buffer)
+{
+    assert(samples >= 0);
+    mp_audio_realloc_min(ab->buffer, ab->buffer->samples + samples);
+    *out_buffer = *ab->buffer;
+    out_buffer->samples = ab->buffer->samples + samples;
+    mp_audio_skip_samples(out_buffer, ab->buffer->samples);
+}
+
+void mp_audio_buffer_finish_write(struct mp_audio_buffer *ab, int samples)
+{
+    assert(samples >= 0 && samples <= mp_audio_buffer_get_write_available(ab));
+    ab->buffer->samples += samples;
+}
+
+// Append data to the end of the buffer.
+// If the buffer is not large enough, it is transparently resized.
+// For now always copies the data.
+void mp_audio_buffer_append(struct mp_audio_buffer *ab, struct mp_audio *mpa)
+{
+    int offset = ab->buffer->samples;
+    ab->buffer->samples += mpa->samples;
+    mp_audio_realloc_min(ab->buffer, ab->buffer->samples);
+    mp_audio_copy(ab->buffer, offset, mpa, 0, mpa->samples);
+}
+
+// Prepend silence to the start of the buffer.
+void mp_audio_buffer_prepend_silence(struct mp_audio_buffer *ab, int samples)
+{
+    assert(samples >= 0);
+    int oldlen = ab->buffer->samples;
+    ab->buffer->samples += samples;
+    mp_audio_realloc_min(ab->buffer, ab->buffer->samples);
+    mp_audio_copy(ab->buffer, samples, ab->buffer, 0, oldlen);
+    mp_audio_fill_silence(ab->buffer, 0, samples);
+}
+
+// Get the start of the current readable buffer.
+void mp_audio_buffer_peek(struct mp_audio_buffer *ab, struct mp_audio *out_mpa)
+{
+    *out_mpa = *ab->buffer;
+}
+
+// Skip leading samples. (Used with mp_audio_buffer_peek() to read data.)
+void mp_audio_buffer_skip(struct mp_audio_buffer *ab, int samples)
+{
+    assert(samples >= 0 && samples <= ab->buffer->samples);
+    mp_audio_copy(ab->buffer, 0, ab->buffer, samples,
+                  ab->buffer->samples - samples);
+    ab->buffer->samples -= samples;
+}
+
+void mp_audio_buffer_clear(struct mp_audio_buffer *ab)
+{
+    ab->buffer->samples = 0;
+}
+
+// Return number of buffered audio samples
+int mp_audio_buffer_samples(struct mp_audio_buffer *ab)
+{
+    return ab->buffer->samples;
+}
+
+// Return amount of buffered audio in seconds.
+double mp_audio_buffer_seconds(struct mp_audio_buffer *ab)
+{
+    return ab->buffer->samples / (double)ab->buffer->rate;
+}
diff --git a/audio/audio_buffer.h b/audio/audio_buffer.h
new file mode 100644
index 0000000000..8cd0df30d0
--- /dev/null
+++ b/audio/audio_buffer.h
@@ -0,0 +1,44 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef MP_AUDIO_BUFFER_H
+#define MP_AUDIO_BUFFER_H
+
+struct mp_audio_buffer;
+struct mp_audio;
+struct mp_chmap;
+
+struct mp_audio_buffer *mp_audio_buffer_create(void *talloc_ctx);
+void mp_audio_buffer_reinit(struct mp_audio_buffer *ab, struct mp_audio *fmt);
+void mp_audio_buffer_reinit_fmt(struct mp_audio_buffer *ab, int format,
+                                const struct mp_chmap *channels, int srate);
+void mp_audio_buffer_get_format(struct mp_audio_buffer *ab,
+                                struct mp_audio *out_fmt);
+void mp_audio_buffer_preallocate_min(struct mp_audio_buffer *ab, int samples);
+int mp_audio_buffer_get_write_available(struct mp_audio_buffer *ab);
+void mp_audio_buffer_get_write_buffer(struct mp_audio_buffer *ab, int minsamples,
+                                      struct mp_audio *out_buffer);
+void mp_audio_buffer_finish_write(struct mp_audio_buffer *ab, int samples);
+void mp_audio_buffer_append(struct mp_audio_buffer *ab, struct mp_audio *mpa);
+void mp_audio_buffer_prepend_silence(struct mp_audio_buffer *ab, int samples);
+void mp_audio_buffer_peek(struct mp_audio_buffer *ab, struct mp_audio *out_mpa);
+void mp_audio_buffer_skip(struct mp_audio_buffer *ab, int samples);
+void mp_audio_buffer_clear(struct mp_audio_buffer *ab);
+int mp_audio_buffer_samples(struct mp_audio_buffer *ab);
+double mp_audio_buffer_seconds(struct mp_audio_buffer *ab);
+
+#endif
diff --git a/audio/decode/ad.h b/audio/decode/ad.h
index 30e739d135..6c76e8dfd0 100644
--- a/audio/decode/ad.h
+++ b/audio/decode/ad.h
@@ -24,6 +24,7 @@
 #include "demux/demux.h"
 
 #include "audio/format.h"
+#include "audio/audio.h"
 
 struct mp_decoder_list;
 
@@ -35,8 +36,7 @@ struct ad_functions {
     int (*init)(sh_audio_t *sh, const char *decoder);
     void (*uninit)(sh_audio_t *sh);
     int (*control)(sh_audio_t *sh, int cmd, void *arg);
-    int (*decode_audio)(sh_audio_t *sh, unsigned char *buffer, int minlen,
-                        int maxlen);
+    int (*decode_audio)(sh_audio_t *sh, struct mp_audio *buffer, int maxlen);
 };
 
 enum ad_ctrl {
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index 72bb51e00c..2e0cade8c5 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -36,25 +36,20 @@
 #include "mpvcore/av_opts.h"
 
 #include "ad.h"
-#include "audio/reorder_ch.h"
 #include "audio/fmt-conversion.h"
 
-#include "compat/mpbswap.h"
 #include "compat/libav.h"
 
 struct priv {
     AVCodecContext *avctx;
     AVFrame *avframe;
-    uint8_t *output;
-    uint8_t *output_packed; // used by deplanarize to store packed audio samples
-    int output_left;
-    int unitsize;
+    struct mp_audio frame;
     bool force_channel_map;
     struct demux_packet *packet;
 };
 
 static void uninit(sh_audio_t *sh);
-static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen);
+static int decode_new_packet(struct sh_audio *sh);
 
 #define OPT_BASE_STRUCT struct MPOpts
 
@@ -138,22 +133,21 @@ static int preinit(sh_audio_t *sh)